Known analog systems exist for processing speech for transmission by radio. One system of this type is known as "LINCOMPEX" and, by means of variable attenuation at the transmitter end, it enables the speech modulation level to be maintained more or less constant for a period of a few tens of milliseconds. This considerably improves the signal/noise ratio obtained at reception. In such systems, the value of the attenuation applied before transmission is itself transmitted via a frequency multiplexed auxiliary channel by frequency modulation of a sub-carrier situated at the top of the telephone band. The modulated sub-carrier is used to control an expander located at the receiver, i.e. a second attenuator having the reverse function of the attenuator used at the transmitter, whereby the speech level prior to compression is restored. Such purely analog systems are used in radiotelephony, in particular for decametric wavelengths.
An improvement to the "LINCOMPEX" system is known under the name "SYNCOMPEX". In the "SYNCOMPEX" system, speech is processed in packets of finite duration (13.33 milliseconds) and an auxiliary channel in the form of an auxiliary radioelectric frequency channel is used to transmit digitally encoded data. The corresponding codes are transmitted by frequency shift keying modulation (FSK modulation). This data is diversity propagated over two sub-carriers both of which are inserted in the voice band. In this system, the speech components existing in the bands used for transmitting the FSK channels are eliminated. Diversity transmission in which the digital data is transmitted over two different channels enables the receiver to choose the channel which has the better signal to noise ratio, and consequently limits transmission errors as much as possible.
However, there remains a problem in using these systems for so-called adaptive algorithms, which, for example, use differential coding for encoding speech in digital form and for taking advantage of the peculiarities of the dynamic range of speech and of the different degrees of sensitivity of the human ear to various kinds of degradation, and for reducing the binary data rate of the speech as transmitted between the transmitter and the receiver, since such arrangements always produce the result that the reduced data rate of coded speech samples increases the sensitivity of the system to errors in transmission and that these errors cannot be corrected by simple processing methods.
The dynamic range of amplitude variations in the speech signal produced by a speaker comprises a syllabic term and a melodic term.
The syllabic term is a high amplitude term having an amplitude of about 30 decibels for some speakers and its frequency band goes from very low frequencies to about 10 Hz. The average syllabic rate is generally about 4 to 7 Hz. In practice, the syllabic term is not significant in speech intelligibility and it is not therefore very important in providing agreeable transmission. However, it is used in transmission to improve listening comfort by eliminating radioelectric noise during silences in speech, and it is also used in conventional telephony for stabilizing amplifier circuits by conserving the equivalent.
The melodic term has a smaller amplitude of 10 to 15 decibels and its frequency band covers the spectrum from about 6 to 250 Hz depending on the speaker. The melodic term is extremely important to hearing cognation and to speaker recognition, and it is therefore essential that its frequency should be accurately reproduced.
Unfortunately, in known devices which perform adaptive digital transmission of speech, the transmission noise contains at least as many harmonic frequencies of the melodic spectrum as signals significant to syllabic adaptation, which causes reception errors that cannot be eliminated by simple processing methods using, for example, analog filters.
An object of the present invention is to remedy the above drawbacks.